Wiley Internet Communications Using SIP Delivering VoIP And Multimedia Services With Session Initiation Protocol 2nd Edition Jul 2006 ISBN 0471776572 pdf
Internet Communications
Using SIP
Delivering VoIP and Multimedia Services
with Session Initiation Protocol
Second Edition
Henry Sinnreich
Alan B. Johnston
Internet Communications
Using SIP
Second Edition
Internet Communications
Using SIP
Delivering VoIP and Multimedia Services
with Session Initiation Protocol
Second Edition
Henry Sinnreich
Alan B. Johnston
Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session
Initiation Protocol, Second Edition
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Library of Congress Cataloging-in-Publication Data
Sinnreich, Henry.
Internet communications using SIP : delivering VoIP and multimedia services with Section Initiation Protocol / Henry Sinnreich, Alan B. Johnston. — 2nd ed.
p. cm.
Includes index.
ISBN-13: 978-0-471-77657-4 (cloth)
ISBN-10: 0-471-77657-2 (cloth)
1. Computer network protocols. 2. Internet telephony. 3. Multimedia systems. I. Title.
TK5105.55.S56 2006
621.3850285’4678—dc22
2006009325
Trademarks: Wiley, the Wiley logo, and related trade dress are trademarks or registered trademarks of John Wiley & Sons, Inc. and/or its affiliates, in the United States and other countries,
and may not be used without written permission. All other trademarks are the property of their
respective owners. Wiley Publishing, Inc., is not associated with any product or vendor mentioned in this book.
Wiley also publishes its books in a variety of electronic formats. Some content that appears in
print may not be available in electronic books.
We could not have written this book without the support of our forgiving
spouses, Fabienne and Lisa, who held the fort while we were working on
SIP. And to both our family members shouting, “Your SIP phone is ringing.”
About the Authors
Dr. Henry Sinnreich (Richardson, TX) is Chief Technology Officer at Pulver.com,
a leading media company for VoIP and Internet communication services. Dr.
Sinnreich has held engineering and executive positions at MCI where he was
an MCI fellow and has been involved in Internet and multimedia services for
more than 12 years, including the development of the flagship MCI Advantage
service based on SIP. Henry Sinnreich is also a contributor to IETF standards
for Internet communications in such areas as SIP telephony devices and using
RTP extensions for voice quality monitoring. He was awarded the title Pioneer
for VoIP in 2000 at the VON Europe conference. Henry Sinnreich has been a
cofounder and board member of the International SIP Forum based in Stockholm. He is a frequent speaker and is known as the leading evangelist, worldwide, for SIP based VoIP, presence, IM, multimedia, and integration of
applications with communications. Dr. Sinnreich is also a guest lecturer at the
Engineering School of the Southern Methodist University in Dallas, TX.
Alan B. Johnston (St. Louis, MO) is a Consulting Member of Technical Staff at
Avaya, Inc. He has coauthored the core Internet SIP standard RFC 3261 and four
other SIP related RFCs. He is the co-chair of the IETF Centralized Conferencing
Working Group and is on the board of directors of the International SIP Forum.
His current areas of interest include peer-to-peer SIP and security. Dr. Johnston
is a frequent speaker and lecturer on SIP and contributor to various publications,
and is an adjunct professor at Washington University in St. Louis, MO.
vii
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ix
Contents
Foreword
xxi
Acknowledgments
Introduction
xxiii
xxv
Chapter 1
Introduction
Problem: Too Many Public Networks
Incompatible Enterprise Communications
Network Consolidation: The Internet
Voice over IP
Presence—The Dial Tone for the Twenty-First Century?
The Value Proposition of SIP
SIP Is Not a Miracle Protocol
The Short History of SIP
References in This Book
SIP Open Source Code and SIP Products
References for Telephony
Summary
References
1
1
4
4
5
6
6
6
7
8
9
10
10
10
Chapter 2
Internet Communications Enabled by SIP
Internet Multimedia Protocols
The Value of Signaling
Protocols for Media Description, Media Transport, and other
Multimedia Delivery
Addressing
SIP in a Nutshell
SIP Capabilities
Overview of Services Provided by SIP Servers
Peer-to-Peer SIP (P2PSIP)
11
12
13
14
15
15
17
18
19
xi
xii
Contents
Caller Preferences
Mobility in the Wider Concept
Global Telephone Number Portability
SIP Application-Level Mobility
Context-Aware Communications: Presence and IM
21
21
23
23
E-Commerce: Customer Relations Management
Conferencing and Collaboration
Telephony Call Control Services
Intelligent Network Services Using SIP: ITU Services CS-1
and CS-2
SIP Service Creation—Telephony-Style
ENUM
SIP Interworking with ITU-T Protocols
Mixed Internet-PSTN Services
23
24
25
SIP Security
SIP Accessibility to Communications for the Hearing and
Speech Disabled
SIP Orphans
Commercial SIP Products
What SIP Does Not Do
Divergent Views on the Network
Summary
References
Architectural Principles of the Internet
Telecom Architecture
Internet Architecture
The Internet Backbone Architecture
The Internet Standards Process
Protocols and Application Programming Interfaces
Is XML the Presentation Layer of the Internet Protocol
Architecture?
Chapter 4
20
20
SIP Presence
Instant Messaging
The Integration of Communications with Applications
PSTN and INTerworking (PINT)
SPIRITS
TRIP
Chapter 3
19
20
25
26
27
27
29
29
29
29
31
31
32
32
33
34
35
35
39
39
42
44
48
49
50
Middle-Age Symptoms of the Internet
Fighting Complexity
Summary
References
50
51
52
52
DNS and ENUM
Introduction
Addressing on the Internet
53
53
54
The Universal Resource Identifier (URI)
mailto:
54
55
Contents
The Universal Resource Locator (URL)
Tel URI
The phone-context
SIP URI
IANA ENUM Service Registrations
The Domain Name System
Delegation
Caching
A Partial DNS Glossary
DNS and ENUM Usage Example
Finding an Outgoing SIP Server
Finding an Incoming SIP Server in the ENUM Case
Call Setup Delay
DNS-Based Routing Service Using SIP
SIP URI or Telephone Number?
The ENUM Functional Architecture
69
ENUM and Number Portability
Implementation Issues
71
71
DNS and SIP User Preferences
Application Scenarios for SIP Service Using ENUM
PBX Enterprise Voice Network
Enterprise System with IP Communications
Residential User with ENUM Service
Miscellaneous: ENUM Lookup of the Display Name
DNS and Security
Impersonation
Eavesdropping
Data Tampering
Malicious Redirection
Denial of Service
Chapter 5
55
56
56
57
58
58
59
59
60
62
63
64
67
67
67
72
73
74
74
76
76
77
77
77
78
78
78
Summary
References
79
79
Real-Time Internet Multimedia
Introduction
Freshening Up on IP
Multicast Protocols
81
81
83
85
Multicast Address Allocation
Application-Level Multicast
85
86
Transport Protocols
IP Network Layer Services
Differentiated Services
Resource Reservation
Integrated Services and DiffServ Networks
Multiprotocol Label Switching
Media and Data Formats
Media Transport Using RTP
RTP Payloads and Payload Format Specifications
86
87
88
88
89
89
90
91
92
xiii
xiv
Contents
Chapter 6
Multimedia Server Recording and Playback Control
Session Description
Session Announcements
Session Invitation
Authentication and Key Distribution
Summary
References
93
93
93
93
94
94
94
SIP Overview
What Makes SIP Special
97
97
SIP Enabled Network
Watching How Sausages Are Being Made
What SIP Is Not
Introduction to SIP
Elements of a SIP Network
User Agents
Servers
Location Services
SIP Functions
Address Resolution
Session-Related Functions
Session Setup
Media Negotiation
Session Modification
Session Termination and Cancellation
Mid-Call Signaling
Call Control
Preconditions Call Setup
Nonsession-Related Functions
Mobility
Message Transport
Event Subscription and Notification
Presence Publication
Authentication Challenges
Extensibility
Chapter 7
98
101
102
102
106
106
106
107
107
108
110
110
111
114
116
117
118
121
123
124
126
127
128
128
130
Summary
References
132
132
SIP Service Creation
Services in SIP
135
135
Service Example
Server Implementation
Called User Agent Implementation
Calling User Agent Implementation
Comparison
New Methods and Headers
Service Creation Options
136
136
137
138
140
141
142
Contents
Call Processing Language
Introduction to CPL
Example of CPL Scripts
SIP Common Gateway Interface
SIP Application Programming Interfaces
SIP Servlets
JAIN
Chapter 8
SIP and VoiceXML
Summary
References
149
150
150
User Preferences
Introduction
Preferences of Caller
153
153
154
Example for Contact
Example for Accept-Contact
Example for Reject-Contact
Chapter 9
142
142
146
147
148
149
149
156
156
156
Preferences of the Called Party
Server Support for User Preferences and for Policies
Summary
References
157
157
157
158
SIP Security
Threats
159
159
Session Setup
Presence and IM
160
161
Security Mechanisms
162
Authentication
Confidentiality
Secure SIP URI Scheme
Integrity
Identity
Media Security
SRTP
MIKEY
SDP Security Descriptions
New Directions
162
163
164
165
165
166
166
167
167
168
DTLS
ZRTP
169
169
Summary
References
169
170
Chapter 10 NAT and Firewall Traversal
Network Address Translators
Firewalls
173
174
177
STUN, TURN, and ICE
Application Layer Gateways
Privacy Considerations
179
180
183
xv
xvi
Contents
Summary
References
184
184
Chapter 11 SIP Telephony
Basic Telephony Services
185
185
SIP and PSTN Interworking
Gateway Location and Routing
SIP/PSTN Protocol Interworking
Types of Gateways
SIP and Early Media
SIP Telephony and ISUP Tunneling
Enhanced Telephony Services
Call Control Services and Third-Party Call Control
Problem Statement
The REFER Method
SIP Third-Party Call Control
Basic Third-Party Call Control
Security for Third-Party Call Control
Peer-to-Peer Third-Party Call Control
Summary
References
Chapter 12 Voicemail and Universal Messaging
Problem Statement for Unified Messaging
Architecture and Operation
RTSP-Enabled Voice Message Retrieval
Depositing of Voice Messages
Notification for Waiting Messages
Simple Message Notification Format
Rich Message Notification Format
Retrieval of Messages
Summary
References
185
186
187
188
188
190
196
199
199
201
202
203
203
205
206
207
209
209
211
212
214
217
217
220
221
221
221
Chapter 13 Presence and Instant Messaging
The Potential of SIP Presence, Events, and IM
The Evolution of IM and Presence
The IETF Model for Presence and IM
Client Server and Peer-to-Peer Presence and IM
SIP Event-Based Communications and Applications
223
224
225
226
228
229
Presence Event Package
Presence Information Data Format
231
233
The Data Model for Presence
Indication of Message Composition for IM
Rich Presence Information
SIP Extensions for Instant Messaging
Summary
References
235
236
236
239
241
242
Contents
Chapter 14 SIP Conferencing
Introduction
SIP Conferencing Models
Ad Hoc and Scheduled Conferences
Changing the Nature of a Conference
Centralized Conferencing
Summary
References
245
245
246
249
249
251
251
251
Chapter 15 SIP Application Level Mobility
Mobility in Different Protocol Layers
Dimensions of Mobility
Examples of SIP Application-Layer Mobility
SIP Network-Based Fixed-Mobile Convergence
SIP Device-Based Fixed-Mobile Convergence
SIP Application-Layer Mobility and Mobile IP
Multimodal Mobile Device Technology and Issues
253
254
255
256
261
263
263
265
Network Control versus User Control of Mobility
IEEE 802.21 Media-Independent Handover (MIH)
Network Selection Issues
266
267
269
Summary
References
Chapter 16 Emergency and Preemption Communication Services
Requirements
Location Information
270
270
273
274
275
Types of Location Information
Sources of Location Information
DNS-Based Location Information
275
275
275
Internet-Based Emergency Calling
277
Identifying an Internet Emergency Call: The SOS URI
Internet Emergency Call Routing
Security for Emergency Call Services
Using the PSTN for VoIP Emergency Calls
Emergency Communication Services
Emergency Call Preemption Using SIP
Linking SIP Preemption to IP Network and Link Layer
Preemption
Summary
References
Chapter 17 Accessibility for the Disabled
About Accessibility
Accessibility on Legacy Networks and on the Internet
Requirements for Accessibility
Text over IP (ToIP)
Performance Metrics for ToIP
278
278
279
280
281
282
284
285
285
287
287
288
289
290
293
xvii
xviii Contents
Transcoding Services
Transcoding Scenarios
Call Control Models for Transcoding Services
Summary
References
Chapter 18 Quality of Service for Real-Time Internet Communications
Voice Quality Metrics
Delay Limits for Voice
Burst vs. Average Packet Loss
Acoustics and the Network
Internet Codecs
Codecs in Wireless Networks and Transcoding
Codec Bandwidth
The Endpoint Quality for Voice
The Internet Performance
294
294
296
298
299
301
303
303
304
304
305
307
307
308
308
Concerns Regarding Congestion Control
Internet Traffic Statistics: Voice Is Negligible
309
309
A Summary of Internet QoS Technologies
Best Effort Is for the Best Reasons
Monitoring QoS for Real-Time Communications
Summary
References
311
313
314
315
315
Chapter 19 SIP Component Services
Master/Slave VoIP Systems
IP Telephony Gateways
The Converged Applications Environment
The Control of Service Context
Voicemail
Collecting DTMF Digits
Interactive Voice Response System
Scheduled Conference Service
Summary
References
Chapter 20 Peer-to-Peer SIP
Definitions for P2P Networks
Overlay Networks
Peer-to-Peer Networks
Distributed Hash Tables (DHTs)
Characteristics of P2P Computing
Security of P2P Networks
The Chord Protocol
P2P SIP
CS SIP Model
P2P SIP Model
317
318
320
323
326
328
330
333
335
337
337
339
340
340
341
342
344
344
345
346
347
348
Contents
Use Cases for P2P SIP
Disruption of the VoIP Infrastructure Model
Summary
References
348
349
350
351
Chapter 21 Conclusions and Future Directions
Short Term Challenges
Future Services: The Internet Is the Service
Still to Develop: Peer-to-Peer SIP Standards
Prediction: The Long Road Ahead
Summary
References
353
355
355
355
356
356
356
Index
357
xix
Foreword
About 10 years ago, the first drafts describing the Session Initiation Protocol
(1996) were published, with the rather modest ambition of setting up multicast
groups for multimedia conferences. In the intervening decade, a draft of about
20 pages has turned into an ecosystem of dozens of RFCs, hundreds of Internet drafts—and several books, conferences, and a magazine. It has become difficult to get a feel for the overall landscape, to distinguish the important core
concepts from the niche applications. This book offers a detailed, technically
informed, yet accessible, introduction to the overall SIP ecosystem, suitable
both for someone who needs to understand the technology to make strategic
decisions and implementers who need to build new components.
SIP is part of the second wave of Internet application protocol. While the
first wave largely focused on asynchronous communications (such as e-mail,
and data transfer), this second wave introduces the notion of interactive,
human-to-human communication that allows integration with any media, not
just voice. As SIP and interactive communications have matured, the goal for
human-to-human communication has shifted. Initially, cell phones promised
voice communication at any time, at any place. Multimedia communications,
on PCs and maybe emerging cellular networks, allow us to add “any media.”
However, the “any time, any place, any media” can also turn us into slaves of
our communications devices, interrupting our ability to think, to eat in peace,
and to meet in person. Thus, our goal has to be to design communications
technology that offers the right media, at the right place, and at the right time.
With some of the advanced functionality of SIP, such as presence, locationbased services, user-created services, and caller preferences, we can get closer
to creating communication systems that support our work and enhance our
personal life.
xxi
xxii
Foreword
With new communications technologies, there is always the temptation to
mimic the old. E-mail inherited aspects of the interoffice memo and fax; web
pages attempted to look like newsprint and brochures. However, in VoIP, there
is the particular temptation to recreate old technology features, as interoperability with the old PSTN will remain important for at least another decade.
Fax-to-email gateways were never quite as important as VoIP-to-PSTN gateways. This emphasis on interoperability with 100-year-old technology has
provided a financial motivation—provide the same service more cheaply.
However, this may also hold back the promise offered by Internet-based multimedia communications, such as the integration of presence, the ability not
just to communicate by voice and maybe video but also to share any application, or the ability to customize the user experience and integrate interactive
communications with existing Internet tools and applications. Just as most
microprocessors are embedded in household appliances and cars, not desktop
PCs and laptops, we might find that Internet-based voice and multimedia
communications will be integrated into games, appliances, and cameras, or be
hidden behind a link on a web page, rather than dialed by name or number. As
for many of the most innovative applications, users will likely not even consider them phone services at all, but extensions that make some other application more productive or more fun.
This book is like a good tour guide to a foreign country. It doesn’t just
describe the major sites and tourist attractions; it lets the reader share in the
history, spirit, language, and culture of the place. Natives write the best tour
guides, and the authors have been living and working in SIP land since it was
a small outpost in one large country called the IETF. The authors have served
as ambassadors in lands near and far, but have also made major contributions
to the development of this part of the Internet landscape, always reminding
others of the original goals of the first inhabitants. After taking the tour, the
reader will be ready not just to show off a stamp on a passport or certificate but
also to contribute to new modes of communications. SIP land is still young and
needs lots of pioneers who can push the frontiers of Internet-enabled communications. There might not always be gold in those hills, but enriching human
communications will always be its own reward.
Henning Schulzrinne
Professor, Columbia University
Acknowledgments
We have enjoyed the benefit of early and significant support from colleagues
and management in MCI. Vint Cerf was, as mentioned, one of the early supporters, and so were Teresa Hastings, John Gallant, Bob Spry, and Robert
Oliver who first took the responsibility for developing and deploying SIP in
their respective engineering departments. John Truetken, Lance Lockhart, and
many other engineers in MCI also had critical contributions to the implementation of SIP. Fred Briggs, Patrice Carroll, Barry Zip, and Leo Cyr from MCI
helped with the challenge to develop marketable services based on SIP. We
were fortunate to work jointly in the development and deployment of SIP services with Steve Donovan, Diana Rawlins, Dean Willis, Robert Sparks, Ben
Campbell, Chris Cunningham, Kevin Summers, and many other engineers
from MCI and elsewhere in the industry engaged in the development of SIP in
the Internet Engineering Task Force (IETF).
Most ideas and inspirations driving SIP are due to Prof. Henning
Schulzrinne from Columbia University and to Jonathan Rosenberg from
DynamicSoft and are reflected in this book. Among the many industry contributors, we gratefully acknowledge discussions and guidance from Rohan
Mahy from Cisco Corporation, Gonzalo Camarillo and Adam Roach from
L.M. Ericsson. Jiri Kuthan from GMD Focus, Berlin, was helpful with SIP tutorial charts and with discussions in transatlantic calls using SIP phones—again,
calls of crystal clear clarity to our surprise. The authors are grateful to Richard
Shockey from NeuStar, Inc. and Douglas Ranalli from NetNumber, Inc. for
numerous discussions regarding ENUM. Theodore Havinis has contributed to
the SIP-QoS-AAA aspect for mobile users.
xxiii
xxiv Acknowledgments
We acknowledge countless helpful discussions and insight from many participants in the IETF and especially to Scott Bradner for holding the authors
and others in the IETF SIP community in line to the true conceptual, technical,
and procedural spirit of the Internet.
Jeff Pulver has played a special role in providing a platform and leading
exhibition of products for what was initially an obscure and unknown protocol in the Voice ON the Net (VON) and other conferences held in America,
Europe, and Asia.
Carrol Long, Kevin Shafer, and Adoabi Obi Tulton from John Wiley & Sons
have been instrumental in editing this book.
Introduction
The second edition of Internet Communications Using SIP had to be rewritten
almost from the ground up, because of the dramatic changes in the industry in
the five years that have passed since the first edition. Some of the developments
had been envisaged in the first edition, but naturally, some have not.
The Internet Has Replaced the Telephone System and
the Telecommunication Networks
Since the publication in 2001 of the first edition of this book, Internet Communications Using SIP, Voice over IP (VoIP) has developed from an emerging technology to the recognized replacement of existing global telephone systems
based on Time Division Multiplex (TDM) circuit switching. The Internet has
also replaced the proposed connection-oriented offsprings of TDM, such as the
Integrated Services Digital Network (ISDN) and the Asynchronous Transfer
Multiplex (ATM) based broadband version BISDN, envisaged for the telecommunications industry by the International Telecommunications Union ITU-T
standards body. TDM, ATM, ISDN, and BISDN are now history.
All wired and wireless communications are instead migrating to the Internet
standards developed by the Internet Engineering Task Force (IETF). The legacy
telecommunication networks, while still dominant, are recognized as a presentday cash cow only and are scheduled for replacement by IP networks.
The end-to-end nature of the Internet that places intelligence in the applications running in the endpoints and gives control to the user at the endpoints
has indeed replaced TDM-based telephony with central control. The Internet
xxv
xxvi Introduction
has also proven to be the home network for other types of communications,
information, entertainment, and data applications. To quote Jon Peterson, area
director of the IETF:
“The Internet is the service.”
The Session Initiation Protocol Is the Standard
for VoIP and Multimedia Communications
Another change from the first edition of this book is the Session Initiation Protocol (SIP), which has been adopted by practically all public VoIP service
providers for wired and wireless communications. The discussions about SIP
versus H.323 standardized by the ITU-T are over as well. The installed base of
H.323 is considered a liability and planned for replacement by SIP sooner or
later.
A global industry has emerged to take advantage of SIP and its associated
IETF standards for real-time communications. More than 560 VoIP service
providers have been reported [1] in early 2006, most of them using SIP-based
networks. The list of SIP-based equipment (such as SIP phones, software for
PCs, and mobile devices, servers, gateways, and so on) is now large and still
growing. Actually, all equipment and system vendors are now supporting SIP.
Presence and Instant Messaging Are
Mainstream Communications
Presence and instant messaging (IM) are now mainstream with consumers
and, in the enterprise, complementing or sometimes replacing voice communications in specific situations (such as in circumstances where silence is
required). Even for VoIP, presence has emerged not only as a valuable
enhancement, but presence may be the dial tone of the twenty-first century.
Presence and event-based communications have enabled the integration of
communications with applications. Presence and IM are discussed in Chapter
13, “Presence and Instant Messaging.”
The so-called IM services provided by large Internet companies, such as
AOL, Apple, Google, IBM, Microsoft, Skype (not SIP-based), and Yahoo!, actually carry at present most of the public VoIP traffic between end users around
the globe.
It is not far-fetched to see the IM Internet companies replacing the former
telephone companies in the voice communication business. Many legacy
telecommunication companies are also using VoIP to replace the internal TDM
voice networks, but their VoIP services may not survive the advanced technologies deployed by the IM Internet companies and the challenge posed by
peer-to-peer (P2P) communications.
Introduction xxvii
Redefining Communications: Mobility, Emergency and
Equal Access for the Disabled
Internet communications have been known not to be dependent on the location on the Internet. Application-level mobility based on SIP is a key component to seamless mobile communications, as discussed in Chapter 15, “SIP
Application Level Mobility.”
Emergency calling services by users in distress using the Internet (such as
911 in the United States or 112 in Europe) are far more powerful and cost less
than the Public Switched Telephone Network (PSTN) based emergency services. Internet-based emergency calling is indeed in the design stage in a number of countries. Chapter 16, “Emergency and Preemption Communication
Services,” discusses Internet-based emergency services.
The multimedia nature of Internet communications gives hearing- and
speech-impaired people the opportunity to fully participate in rich communications for work and in personal life. Chapter 17, “Accessibility for the Disabled,” discusses access to communications for disabled people.
The Rise of Peer-to-Peer Communications
P2P traffic has risen in the Internet since around 2000 and became the dominant part of Internet traffic by 2004. Since 2004, Skype (which is based on P2P
VoIP, IM, and presence) has also become by far the dominant VoIP provider
worldwide. Since P2P SIP standards work is just emerging as of this writing,
Skype can be considered a prestandard P2P Internet communication service.
The reasons for the emergence of overlay networks and P2P applications
and their nature are discussed in Chapter 20, “Peer-to-Peer SIP,” and also in
Chapter 6, “SIP Overview.” Though the present VoIP industry is built on
client-server (CS) SIP, this may significantly change. To quote David Bryan
from p2p.org:
“P2P SIP may change VoIP to the same extent that VoIP has changed telecommunications.”
VoIP and Multimedia Communications Services Are Still
Fragmented
In spite of all the technological progress, VoIP, IM, presence, and multimedia
services are still a highly fragmented industry:
■■
Telephone services based on VoIP operate as islands and can interconnect (as of this writing) using mostly the legacy Public Switched Telephone Network (PSTN). The service model is giving broadband users
xxviii Introduction
access to the legacy telephone system, actually a voice gateway service
between the Internet and TDM. The business model of most VoIP service providers is just lower cost for legacy-style telephone service, also
called PSTN over IP. The PSTN gateway services are using IP inside
their networks, but users are not exposed to the rich IP services, except
when all parties are on the same network.
■■
The most successful public voice, IM, and presence service is Skype,
which is not standards-based.
■■
Walled gardens: The fragmentation of communications is still actively
pursued by most mobile service providers by deploying systems where
their users can get rich IP multimedia services only on their own networks. The fees to communicate between mobile service providers are a
significant part of the business model, and open connectivity to the
Internet (“Internet neutrality”) is still a hotly debated issue. Internet
neutrality is also still debated by many broadband Internet access
providers (such as DSL and cable companies), although we believe that
enlightened government regulators in the developed countries will
weigh in favor of users and open network access in general.
The proliferation of islands for communications makes them less useful the
more there are, since this proliferation is in denial of Metcalf’s law that the
value of a network increases with the square of the number of points attached
to the network. The Internet with more than 1 billion attached endpoints has
thus the highest value for communications. By contrast, the mobile phone
industry boasts 3 billion users, but in many fragmented networks.
Past Obsessions and Present Dangers: QoS and Security
Network-based quality of service (QoS) for voice and the reliability of the
legacy telephone network have long been used by telephone industry marketers to scare users away from VoIP. In the meantime, all public VoIP services
have proven that Internet best-effort QoS works just fine, as long network congestion is avoided. Internet-based voice can actually be much better than the
3.1 kHz voice over the PSTN. As for reliability, all recent major man-made and
natural disasters have proven the Internet and VoIP to be more resilient than
the existing wireline and wireless telephone networks.
Chapter 18, “Quality of Service for Real-Time Internet Communications,” is
aimed at a balanced approach for QoS, and Chapter 16, “Emergency and Preemption Communication Services,” discusses the Emergency Services based
on SIP.
Introduction xxix
The security threats on the Internet have provided well-justified concerns
about the security of VoIP, and even more, the security of IM. As a result, a new
industry niche, that of VoIP and IM security, has sprung up and, as usual, marketers are first drumming up the vulnerabilities of Internet communications to
prepare the sell for all kinds of security products. Though no significant security breaks have been reported so far for Internet communications, security for
VoIP and IM is still work in progress. Chapter 9, “SIP Security,” deals with SIP
security.
References
[1] A list of VoIP companies is provided at www.myvoipprovider.com.
CHAPTER
1
Introduction
The telecommunications, television, and information technology (IT) network
industries are all transformed by the Internet. The transformation is driven by
the need for growth based on new services, more complete global coverage,
and consolidation. In this chapter, we will explore some of the problems and
solutions for end users and every type of business because of the profound
disruptions caused by the Internet.
Problem: Too Many Public Networks
Before the emergence of the Internet, users and service providers were generally accustomed to thinking in terms of four distinct network types: Networks
for IT (data), networks for voice, mobile networks, and networks for television. Each of these dedicated network types could, in turn, be divided into
many incompatible regional and even country-specific flavors with different
protocol variants.
Thus, we find many types of telephony numbering plans, signaling, and
audio encodings; several TV standards; and various types and flavors of what
the telecom industry calls data networks—all of them incompatible and impossible to integrate into one single global network.
1
2
Chapter 1
The mobile telephone networks have converged on a smaller number
of standards in the second generation (2G) networks and in the emerging third
generation (3G) mobile networks. It may turn out, however, that with the
proliferation of new radio technologies for the so-called 4th generation (4G),
such as Wi-Fi and WiMAX, all modern mobile networks will become just a
wireless access mechanism to the Internet, where all public communications,
entertainment, and applications will reside anyhow.
Data networks that originated in the telecom industry came in many forms,
such as digital private lines, X.25, Integrated Services Digital Network (ISDN),
Switched Multimegabit Data Service (SMDS), Frame Relay, and Asynchronous
Transfer Mode (ATM) networks. These so-called data networks were mostly
inspired by circuit-switched telephony concepts. Their names are meant to
suggest that they were not designed primarily to carry voice.
Voice networks are still used for data and fax because of their general availability, though less and less so. However, these networks have come to the end
of their evolution, since they are fundamentally optimized for voice only. TV
networks were designed and optimized for the distribution of entertainment
video streams.
Needless to say, all network types (data, voice, TV, and mobile) have specific
end-user devices that cannot be ported to other service providers or network
types, and most often cannot be globally deployed.
The impact of the Internet has made the wired and wireless phone companies and the TV cable companies look for new business models that can take
advantage of Internet technologies and protocols, among them the Session Initiation Protocol (SIP) for real-time communications, such as Voice over IP
(VoIP), instant messaging (IM), video, conferencing/collaboration, and others.
Examples of the various categories and their business models are illustrated in
Table 1.1. We assume that most readers are familiar with the acronyms used in
the table, and we also explain these acronyms and terms in the book. They can
also be found in the index.
Table 1.1
Internet Communications in 2005 with Examples from North America
CATEGORY
WHO
PROTOCOLS
Open IM services Pulver FWD, Standard SIP
with VoIP voice
Gizmo/
(competing islands) SIPphone,
Damaka,
Ineen
STRENGTHS
WEAKNESSES
Internet
Presence
Video
User gets SIP
URI
On Net is free
Limited
financing
Introduction
CATEGORY
WHO
PROTOCOLS
STRENGTHS
WEAKNESSES
Closed IM islands
with VoIP
Yahoo, MSN, SIP or other
Google, AOL,
Skype
(the most
innovative)
Internet
Nonstandard
Presence
Walled gardens
Video
On Net is free
PSTN gateways
PSTN over IP
Most “VoIP” SIP
companies
Internet
Anywhere
Video (Packet8)
On net is free
Low-cost PSTN
No new services
Compete on
price
Costly
infrastructure
Telephony
over cable
TV cable
companies
Everything from
PSTN to MGCP
to SIP with “P-”
extensions
Broadband
Internet
Access to
80%+
households
Large
investments in
PSTN and older
VoIP flavors
Wireless walled
gardens
3G mobile
operators
SIP for IMS
with “P-”
extensions
Strong
financing
Wireline emulation Wireline
of IMS: TISPAN
phone
companies
“NGN”
SIP with
“P-”
extensions
Central control
inhibits
innovation
IP network cost
Duplicate IMS &
NGN services
The proliferation of isolated communication islands as shown in Table 1.1
makes them less useful as their number keeps increasing (think of many more
communication islands all over the world). Building communication islands
(also called “walled gardens”) is in conflict with Metcalfe’s law that the value
of the network increases by the square of the number of connected endpoints.
Last, but not least, in case of an emergency, having many networks that cannot
communicate directly is not very helpful.
Closed networks are an impediment for innovation, since innovators must
work (technology and legal agreements) with every closed network separately
to bring a new service or product to market. By contrast, the Internet extends
the reach for new applications and services instantly to the whole world.
3
4
Chapter 1
Another observation from Table 1.1 is that the strongest financing available
is at present for closed networks (walled gardens), the ones that are most limited in reach and usefulness. This raises business issues and regulatory questions (what are the public interest obligations, if any?) that are beyond the
scope of this book.
Incompatible Enterprise Communications
Enterprise communication systems are often an even greater mix of incompatible and disjoint systems and devices:
■■
Proprietary PBX and their phones. Phones from one PBX cannot be
used by another.
■■
Instant messaging is a separate system from the PBX.
■■
Various IM systems don’t talk to each other.
■■
Voice conferencing and web-based collaboration use yet other systems.
Maintaining various incompatible and nonintegrated proprietary enterprise
systems is quite costly and reduces the overall productivity of the workforce.
Network Consolidation: The Internet
The Internet has benefited from a number of different fundamentals compared
to legacy networks, such as the tremendous progress of computing technology
and the open standard Internet protocols that define it. This progress can be
attributed to the expertise of the research, academic, and engineering communities whose dedication to excellence and open collaboration on a global basis
have surpassed the usual commercial pressure for time-to-market and competitive secrecy.
The result is an Internet that uses consistent protocols on a global basis, and is
equally well suited to carry data, transactions, and real-time communications,
such as instant messaging (IM), voice, video, and conferencing/collaboration.
Actually, the Internet is the “dumb network,” designed for any application,
even those not yet invented. This is in stark contrast to the isolated “walled
gardens” with central control of all services illustrated in Table 1.1.
Introduction
Voice over IP
Although the Internet has quickly established itself as the preeminent network
for data, commercial transactions, and audio-video distribution, the use of
voice over the Internet has been slower to develop. This has less to do with the
capability of the Internet to carry voice with equal or higher quality than the
telephone network but rather with the complex nature of signaling in voice
services, as you will see in Chapter 6, “SIP Overview.”
There are various approaches for voice services over the Internet, based on
different signaling and control design. Some examples include the following:
■■
Use signaling concepts from the telephone industry—H.323, MGCP,
MEGACO/H.248.
■■
Use control concepts from the telephone industry—central control and
softswitches.
■■
Use the Internet-centric protocol—Session Initiation Protocol (SIP), the
topic of this book.
The movement from such concepts as telephony call models to discovery/rendezvous and session setup between any processes on any platform
anywhere on the Internet is opening up completely new types of communication services.
The use of SIP for establishing voice, video, and data sessions places telephony as just another application on the Internet, using similar addressing,
data types, software, protocols, and security as found, for example, on the
World Wide Web or e-mail.
Separate networks for voice are no longer necessary, and this is of great consequence for all wired and wireless telephone companies.
Complete integration of voice with all other Internet services and applications probably provides the greatest opportunity for innovation. The open and
distributed nature of this service and the “dumb” network model will
empower many innovators, similar to what has happened with other industries on the Internet and the resulting online economy.
Most IM systems on the Internet already have voice and telephony capability as well, though if it is proprietary, they cannot intercommunicate without
IM gateways, although IM gateways inevitably cannot translate all the
features from one system to another. IM gateways are also transitory in nature,
5
6
Chapter 1
since any changes to a proprietary IM protocol may render the gateway close
to useless. By contrast, SIP-based communications offer a global standardsbased approach for interoperability for presence, IM, voice, and video, as we
will show in the following chapters.
Presence—The Dial Tone for the Twenty-First
Century?
Unsuccessful telephone calls are a serious drag on productivity and a source of
frustration, since both parties waste time and talk to voicemail instead to each
other. Also, the timing of the phone call may not be appropriate or not reach
the called party in a suitable location. The advent of presence, so well-known
from IM systems, can provide much more rich information before trying to
make a call in the first place, compared to just hearing the dial tone. Another
convenience of SIP and presence is that many contact addresses may reside
beneath a buddy icon, so the caller need not to know or worry about picking
the right phone number or URI. Presence may, therefore, replace the dial tone
used in telephony for well over 100 years.
The Value Proposition of SIP
SIP is not just another protocol. SIP redefines communications and is impacting
the telecom industry to a similar or greater degree than other industries. This has
been recognized by all telecom service providers and their vendors for wired
and wireless services, as well as by all IT vendors. Chapter 2 will provide an
overview of how the Internet and SIP are redefining communications.
SIP Is Not a Miracle Protocol
As discussed in Chapter 2, “Internet Communications Enabled by SIP,” SIP is
not a miracle protocol and is not designed to do more than discover remote users
and establish interactive communication sessions. SIP is not meant to ensure
quality of service (QoS) all by itself or to transfer large amounts of data. It is not
applicable for conference floor control. Neither is it meant to replace all known
telephony features, many of which are caused by the limitations of circuitswitched voice or to the regulation of voice services. And such a list can go on.
Various other Internet protocols are better suited for other functions. As for
legacy telephony, not all telephone network features lend themselves to replication on the Internet.
Introduction
The Short History of SIP [1]
By 1996, the Internet Engineering Task Force (IETF) had already developed the
basics for multimedia on the Internet (see Chapter 14, “SIP Conferencing”) in
the Multi-Party, Multimedia Working Group. Two proposals, the Simple Conference Invitation Protocol (SCIP) by Henning Schulzrinne and the Session Initiation Protocol (SIP) by Mark Handley, were announced and later merged to
form Session Initiation Protocol. The new protocol also preserved the HTTP
orientation from the initial SCIP proposal that later proved to be crucial to the
merging of IP communications on the Internet.
Schulzrinne focused on the continuing development of SIP with the objective of “re-engineering the telephone system from ground up,” an “opportunity that appears only once in 100 years,” as we heard him argue at a time
when few believed this was practical.
SIP was initially approved as RFC [2] number 2543 in the IETF in March
1999. Because of the tremendous interest and the increasing number of contributions to SIP, a separate SIP Working Group (WG) was formed in September
1999. The SIP for Instant Messaging and Presence Leveraging (SIMPLE) was
formed in March 2001, followed by SIPPING for applications and their extensions in 2002. The specific needs of SIP developers and service providers have
led to an increasing number of new working groups. This very large body of
work attests both to the creativity of the Internet communications engineering
community, and also to the vigor of the newly created industry.
We will shorten the narrative on the history of SIP by listing the related
working groups (WG) in chronological order in Table 1.2. We have listed for
simplicity the year of the first RFC published by the WG, though the WG was
sometimes formed one to two years earlier. Years denote a new WG that has
not yet produced any RFC.
Table 1.2
History of SIP-Related Working Groups
NAME
FIRST RFC
CHARTER
avt
1996
Real-time transmission of audio and video over
UDP/IP: RTP
mmusic
1998
Internet conferencing and multimedia
communications: SIP, SDP, RTSP
iptel
2000
Routing and call processing for IP telephony: TRIP,
CPL, tel URI
sip
2000
Development of the SIP protocol: SIP methods,
messages, events, URI
(continued)
7
8
Chapter 1
Table 1-2
(continued)
NAME
FIRST RFC
CHARTER
enum
2000
DNS-based use of ITU-T E.164 telephone numbers
sipping
2002
Applications and extensions to SIP
simple
2004
Use of SIP for Instant Messaging (IM) and
Presence
xcon
2005
Centralized conferences
behave
(2005)
Behavior for Network Address Translation (NAT) for
use with SIP, RTP
ecrit
(2005)
Emergency communications (such as 911, 112)
p2psip
(2005)
Peer-to-peer SIP (not yet a formal WG)
The growth of SIP-related standards in the IETF is illustrated and discussed
in Chapter 21, “Conclusions and Future Directions.”
References in This Book
Because of the multiple developments on the Internet, SIP is being used in
ever-more services, user software, and various user devices (such as in SIP
phones, PCs, laptops, PDAs, and mobile phones). This is, in effect, a new
industry and its participants keep making new contributions to the core SIP
standards, mainly in the area of new services and new applications. This book
reflects SIP developments up to and including the 64th IETF in November
2005.
We have included, by necessity, many Internet drafts that are designated
work in progress, since they are the only reference source for this particular
information. Some of these drafts may become standards by the time you are
ready to use them; some may be a work in progress and have a higher version
number than quoted as of this writing; and still others may be found only in an
archive for expired drafts.
The SIP WG drafts that are work in progress can be found online at the IETF
web site:
http://ietf.org/html.charters/sip-charter.html
Additional individual submissions and Internet drafts from other working
groups can be found at the following site:
http://ietf.org/ID.html
Introduction
SIP-related drafts that have expired (older than six months) can be found on
several archives. As of this writing, following are some of the sites:
www.cs.columbia.edu/sip/drafts
www.softarmor.com/sipwg
Readers may also perform a web search, such as Google, for any IETF SIPrelated topic or for any Internet draft or RFC.
Several books have been published on Internet multimedia, Voice over IP,
and SIP, some of which are listed here. [3], [4], [5] They focus mainly on how
SIP works. This book is less about explaining how SIP works, but rather what
it does and the new communications and services it enables.
We have reproduced some of the exciting services and features discussed in
the IETF SIP WG and its main offsprings, the SIPPING and SIMPLE Working
Groups. Also in
Using SIP
Delivering VoIP and Multimedia Services
with Session Initiation Protocol
Second Edition
Henry Sinnreich
Alan B. Johnston
Internet Communications
Using SIP
Second Edition
Internet Communications
Using SIP
Delivering VoIP and Multimedia Services
with Session Initiation Protocol
Second Edition
Henry Sinnreich
Alan B. Johnston
Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session
Initiation Protocol, Second Edition
Published by
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Copyright © 2006 by Wiley Publishing, Inc., Indianapolis, Indiana
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ISBN-13: 978-0-471-77657-4
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Library of Congress Cataloging-in-Publication Data
Sinnreich, Henry.
Internet communications using SIP : delivering VoIP and multimedia services with Section Initiation Protocol / Henry Sinnreich, Alan B. Johnston. — 2nd ed.
p. cm.
Includes index.
ISBN-13: 978-0-471-77657-4 (cloth)
ISBN-10: 0-471-77657-2 (cloth)
1. Computer network protocols. 2. Internet telephony. 3. Multimedia systems. I. Title.
TK5105.55.S56 2006
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2006009325
Trademarks: Wiley, the Wiley logo, and related trade dress are trademarks or registered trademarks of John Wiley & Sons, Inc. and/or its affiliates, in the United States and other countries,
and may not be used without written permission. All other trademarks are the property of their
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Wiley also publishes its books in a variety of electronic formats. Some content that appears in
print may not be available in electronic books.
We could not have written this book without the support of our forgiving
spouses, Fabienne and Lisa, who held the fort while we were working on
SIP. And to both our family members shouting, “Your SIP phone is ringing.”
About the Authors
Dr. Henry Sinnreich (Richardson, TX) is Chief Technology Officer at Pulver.com,
a leading media company for VoIP and Internet communication services. Dr.
Sinnreich has held engineering and executive positions at MCI where he was
an MCI fellow and has been involved in Internet and multimedia services for
more than 12 years, including the development of the flagship MCI Advantage
service based on SIP. Henry Sinnreich is also a contributor to IETF standards
for Internet communications in such areas as SIP telephony devices and using
RTP extensions for voice quality monitoring. He was awarded the title Pioneer
for VoIP in 2000 at the VON Europe conference. Henry Sinnreich has been a
cofounder and board member of the International SIP Forum based in Stockholm. He is a frequent speaker and is known as the leading evangelist, worldwide, for SIP based VoIP, presence, IM, multimedia, and integration of
applications with communications. Dr. Sinnreich is also a guest lecturer at the
Engineering School of the Southern Methodist University in Dallas, TX.
Alan B. Johnston (St. Louis, MO) is a Consulting Member of Technical Staff at
Avaya, Inc. He has coauthored the core Internet SIP standard RFC 3261 and four
other SIP related RFCs. He is the co-chair of the IETF Centralized Conferencing
Working Group and is on the board of directors of the International SIP Forum.
His current areas of interest include peer-to-peer SIP and security. Dr. Johnston
is a frequent speaker and lecturer on SIP and contributor to various publications,
and is an adjunct professor at Washington University in St. Louis, MO.
vii
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ix
Contents
Foreword
xxi
Acknowledgments
Introduction
xxiii
xxv
Chapter 1
Introduction
Problem: Too Many Public Networks
Incompatible Enterprise Communications
Network Consolidation: The Internet
Voice over IP
Presence—The Dial Tone for the Twenty-First Century?
The Value Proposition of SIP
SIP Is Not a Miracle Protocol
The Short History of SIP
References in This Book
SIP Open Source Code and SIP Products
References for Telephony
Summary
References
1
1
4
4
5
6
6
6
7
8
9
10
10
10
Chapter 2
Internet Communications Enabled by SIP
Internet Multimedia Protocols
The Value of Signaling
Protocols for Media Description, Media Transport, and other
Multimedia Delivery
Addressing
SIP in a Nutshell
SIP Capabilities
Overview of Services Provided by SIP Servers
Peer-to-Peer SIP (P2PSIP)
11
12
13
14
15
15
17
18
19
xi
xii
Contents
Caller Preferences
Mobility in the Wider Concept
Global Telephone Number Portability
SIP Application-Level Mobility
Context-Aware Communications: Presence and IM
21
21
23
23
E-Commerce: Customer Relations Management
Conferencing and Collaboration
Telephony Call Control Services
Intelligent Network Services Using SIP: ITU Services CS-1
and CS-2
SIP Service Creation—Telephony-Style
ENUM
SIP Interworking with ITU-T Protocols
Mixed Internet-PSTN Services
23
24
25
SIP Security
SIP Accessibility to Communications for the Hearing and
Speech Disabled
SIP Orphans
Commercial SIP Products
What SIP Does Not Do
Divergent Views on the Network
Summary
References
Architectural Principles of the Internet
Telecom Architecture
Internet Architecture
The Internet Backbone Architecture
The Internet Standards Process
Protocols and Application Programming Interfaces
Is XML the Presentation Layer of the Internet Protocol
Architecture?
Chapter 4
20
20
SIP Presence
Instant Messaging
The Integration of Communications with Applications
PSTN and INTerworking (PINT)
SPIRITS
TRIP
Chapter 3
19
20
25
26
27
27
29
29
29
29
31
31
32
32
33
34
35
35
39
39
42
44
48
49
50
Middle-Age Symptoms of the Internet
Fighting Complexity
Summary
References
50
51
52
52
DNS and ENUM
Introduction
Addressing on the Internet
53
53
54
The Universal Resource Identifier (URI)
mailto:
54
55
Contents
The Universal Resource Locator (URL)
Tel URI
The phone-context
SIP URI
IANA ENUM Service Registrations
The Domain Name System
Delegation
Caching
A Partial DNS Glossary
DNS and ENUM Usage Example
Finding an Outgoing SIP Server
Finding an Incoming SIP Server in the ENUM Case
Call Setup Delay
DNS-Based Routing Service Using SIP
SIP URI or Telephone Number?
The ENUM Functional Architecture
69
ENUM and Number Portability
Implementation Issues
71
71
DNS and SIP User Preferences
Application Scenarios for SIP Service Using ENUM
PBX Enterprise Voice Network
Enterprise System with IP Communications
Residential User with ENUM Service
Miscellaneous: ENUM Lookup of the Display Name
DNS and Security
Impersonation
Eavesdropping
Data Tampering
Malicious Redirection
Denial of Service
Chapter 5
55
56
56
57
58
58
59
59
60
62
63
64
67
67
67
72
73
74
74
76
76
77
77
77
78
78
78
Summary
References
79
79
Real-Time Internet Multimedia
Introduction
Freshening Up on IP
Multicast Protocols
81
81
83
85
Multicast Address Allocation
Application-Level Multicast
85
86
Transport Protocols
IP Network Layer Services
Differentiated Services
Resource Reservation
Integrated Services and DiffServ Networks
Multiprotocol Label Switching
Media and Data Formats
Media Transport Using RTP
RTP Payloads and Payload Format Specifications
86
87
88
88
89
89
90
91
92
xiii
xiv
Contents
Chapter 6
Multimedia Server Recording and Playback Control
Session Description
Session Announcements
Session Invitation
Authentication and Key Distribution
Summary
References
93
93
93
93
94
94
94
SIP Overview
What Makes SIP Special
97
97
SIP Enabled Network
Watching How Sausages Are Being Made
What SIP Is Not
Introduction to SIP
Elements of a SIP Network
User Agents
Servers
Location Services
SIP Functions
Address Resolution
Session-Related Functions
Session Setup
Media Negotiation
Session Modification
Session Termination and Cancellation
Mid-Call Signaling
Call Control
Preconditions Call Setup
Nonsession-Related Functions
Mobility
Message Transport
Event Subscription and Notification
Presence Publication
Authentication Challenges
Extensibility
Chapter 7
98
101
102
102
106
106
106
107
107
108
110
110
111
114
116
117
118
121
123
124
126
127
128
128
130
Summary
References
132
132
SIP Service Creation
Services in SIP
135
135
Service Example
Server Implementation
Called User Agent Implementation
Calling User Agent Implementation
Comparison
New Methods and Headers
Service Creation Options
136
136
137
138
140
141
142
Contents
Call Processing Language
Introduction to CPL
Example of CPL Scripts
SIP Common Gateway Interface
SIP Application Programming Interfaces
SIP Servlets
JAIN
Chapter 8
SIP and VoiceXML
Summary
References
149
150
150
User Preferences
Introduction
Preferences of Caller
153
153
154
Example for Contact
Example for Accept-Contact
Example for Reject-Contact
Chapter 9
142
142
146
147
148
149
149
156
156
156
Preferences of the Called Party
Server Support for User Preferences and for Policies
Summary
References
157
157
157
158
SIP Security
Threats
159
159
Session Setup
Presence and IM
160
161
Security Mechanisms
162
Authentication
Confidentiality
Secure SIP URI Scheme
Integrity
Identity
Media Security
SRTP
MIKEY
SDP Security Descriptions
New Directions
162
163
164
165
165
166
166
167
167
168
DTLS
ZRTP
169
169
Summary
References
169
170
Chapter 10 NAT and Firewall Traversal
Network Address Translators
Firewalls
173
174
177
STUN, TURN, and ICE
Application Layer Gateways
Privacy Considerations
179
180
183
xv
xvi
Contents
Summary
References
184
184
Chapter 11 SIP Telephony
Basic Telephony Services
185
185
SIP and PSTN Interworking
Gateway Location and Routing
SIP/PSTN Protocol Interworking
Types of Gateways
SIP and Early Media
SIP Telephony and ISUP Tunneling
Enhanced Telephony Services
Call Control Services and Third-Party Call Control
Problem Statement
The REFER Method
SIP Third-Party Call Control
Basic Third-Party Call Control
Security for Third-Party Call Control
Peer-to-Peer Third-Party Call Control
Summary
References
Chapter 12 Voicemail and Universal Messaging
Problem Statement for Unified Messaging
Architecture and Operation
RTSP-Enabled Voice Message Retrieval
Depositing of Voice Messages
Notification for Waiting Messages
Simple Message Notification Format
Rich Message Notification Format
Retrieval of Messages
Summary
References
185
186
187
188
188
190
196
199
199
201
202
203
203
205
206
207
209
209
211
212
214
217
217
220
221
221
221
Chapter 13 Presence and Instant Messaging
The Potential of SIP Presence, Events, and IM
The Evolution of IM and Presence
The IETF Model for Presence and IM
Client Server and Peer-to-Peer Presence and IM
SIP Event-Based Communications and Applications
223
224
225
226
228
229
Presence Event Package
Presence Information Data Format
231
233
The Data Model for Presence
Indication of Message Composition for IM
Rich Presence Information
SIP Extensions for Instant Messaging
Summary
References
235
236
236
239
241
242
Contents
Chapter 14 SIP Conferencing
Introduction
SIP Conferencing Models
Ad Hoc and Scheduled Conferences
Changing the Nature of a Conference
Centralized Conferencing
Summary
References
245
245
246
249
249
251
251
251
Chapter 15 SIP Application Level Mobility
Mobility in Different Protocol Layers
Dimensions of Mobility
Examples of SIP Application-Layer Mobility
SIP Network-Based Fixed-Mobile Convergence
SIP Device-Based Fixed-Mobile Convergence
SIP Application-Layer Mobility and Mobile IP
Multimodal Mobile Device Technology and Issues
253
254
255
256
261
263
263
265
Network Control versus User Control of Mobility
IEEE 802.21 Media-Independent Handover (MIH)
Network Selection Issues
266
267
269
Summary
References
Chapter 16 Emergency and Preemption Communication Services
Requirements
Location Information
270
270
273
274
275
Types of Location Information
Sources of Location Information
DNS-Based Location Information
275
275
275
Internet-Based Emergency Calling
277
Identifying an Internet Emergency Call: The SOS URI
Internet Emergency Call Routing
Security for Emergency Call Services
Using the PSTN for VoIP Emergency Calls
Emergency Communication Services
Emergency Call Preemption Using SIP
Linking SIP Preemption to IP Network and Link Layer
Preemption
Summary
References
Chapter 17 Accessibility for the Disabled
About Accessibility
Accessibility on Legacy Networks and on the Internet
Requirements for Accessibility
Text over IP (ToIP)
Performance Metrics for ToIP
278
278
279
280
281
282
284
285
285
287
287
288
289
290
293
xvii
xviii Contents
Transcoding Services
Transcoding Scenarios
Call Control Models for Transcoding Services
Summary
References
Chapter 18 Quality of Service for Real-Time Internet Communications
Voice Quality Metrics
Delay Limits for Voice
Burst vs. Average Packet Loss
Acoustics and the Network
Internet Codecs
Codecs in Wireless Networks and Transcoding
Codec Bandwidth
The Endpoint Quality for Voice
The Internet Performance
294
294
296
298
299
301
303
303
304
304
305
307
307
308
308
Concerns Regarding Congestion Control
Internet Traffic Statistics: Voice Is Negligible
309
309
A Summary of Internet QoS Technologies
Best Effort Is for the Best Reasons
Monitoring QoS for Real-Time Communications
Summary
References
311
313
314
315
315
Chapter 19 SIP Component Services
Master/Slave VoIP Systems
IP Telephony Gateways
The Converged Applications Environment
The Control of Service Context
Voicemail
Collecting DTMF Digits
Interactive Voice Response System
Scheduled Conference Service
Summary
References
Chapter 20 Peer-to-Peer SIP
Definitions for P2P Networks
Overlay Networks
Peer-to-Peer Networks
Distributed Hash Tables (DHTs)
Characteristics of P2P Computing
Security of P2P Networks
The Chord Protocol
P2P SIP
CS SIP Model
P2P SIP Model
317
318
320
323
326
328
330
333
335
337
337
339
340
340
341
342
344
344
345
346
347
348
Contents
Use Cases for P2P SIP
Disruption of the VoIP Infrastructure Model
Summary
References
348
349
350
351
Chapter 21 Conclusions and Future Directions
Short Term Challenges
Future Services: The Internet Is the Service
Still to Develop: Peer-to-Peer SIP Standards
Prediction: The Long Road Ahead
Summary
References
353
355
355
355
356
356
356
Index
357
xix
Foreword
About 10 years ago, the first drafts describing the Session Initiation Protocol
(1996) were published, with the rather modest ambition of setting up multicast
groups for multimedia conferences. In the intervening decade, a draft of about
20 pages has turned into an ecosystem of dozens of RFCs, hundreds of Internet drafts—and several books, conferences, and a magazine. It has become difficult to get a feel for the overall landscape, to distinguish the important core
concepts from the niche applications. This book offers a detailed, technically
informed, yet accessible, introduction to the overall SIP ecosystem, suitable
both for someone who needs to understand the technology to make strategic
decisions and implementers who need to build new components.
SIP is part of the second wave of Internet application protocol. While the
first wave largely focused on asynchronous communications (such as e-mail,
and data transfer), this second wave introduces the notion of interactive,
human-to-human communication that allows integration with any media, not
just voice. As SIP and interactive communications have matured, the goal for
human-to-human communication has shifted. Initially, cell phones promised
voice communication at any time, at any place. Multimedia communications,
on PCs and maybe emerging cellular networks, allow us to add “any media.”
However, the “any time, any place, any media” can also turn us into slaves of
our communications devices, interrupting our ability to think, to eat in peace,
and to meet in person. Thus, our goal has to be to design communications
technology that offers the right media, at the right place, and at the right time.
With some of the advanced functionality of SIP, such as presence, locationbased services, user-created services, and caller preferences, we can get closer
to creating communication systems that support our work and enhance our
personal life.
xxi
xxii
Foreword
With new communications technologies, there is always the temptation to
mimic the old. E-mail inherited aspects of the interoffice memo and fax; web
pages attempted to look like newsprint and brochures. However, in VoIP, there
is the particular temptation to recreate old technology features, as interoperability with the old PSTN will remain important for at least another decade.
Fax-to-email gateways were never quite as important as VoIP-to-PSTN gateways. This emphasis on interoperability with 100-year-old technology has
provided a financial motivation—provide the same service more cheaply.
However, this may also hold back the promise offered by Internet-based multimedia communications, such as the integration of presence, the ability not
just to communicate by voice and maybe video but also to share any application, or the ability to customize the user experience and integrate interactive
communications with existing Internet tools and applications. Just as most
microprocessors are embedded in household appliances and cars, not desktop
PCs and laptops, we might find that Internet-based voice and multimedia
communications will be integrated into games, appliances, and cameras, or be
hidden behind a link on a web page, rather than dialed by name or number. As
for many of the most innovative applications, users will likely not even consider them phone services at all, but extensions that make some other application more productive or more fun.
This book is like a good tour guide to a foreign country. It doesn’t just
describe the major sites and tourist attractions; it lets the reader share in the
history, spirit, language, and culture of the place. Natives write the best tour
guides, and the authors have been living and working in SIP land since it was
a small outpost in one large country called the IETF. The authors have served
as ambassadors in lands near and far, but have also made major contributions
to the development of this part of the Internet landscape, always reminding
others of the original goals of the first inhabitants. After taking the tour, the
reader will be ready not just to show off a stamp on a passport or certificate but
also to contribute to new modes of communications. SIP land is still young and
needs lots of pioneers who can push the frontiers of Internet-enabled communications. There might not always be gold in those hills, but enriching human
communications will always be its own reward.
Henning Schulzrinne
Professor, Columbia University
Acknowledgments
We have enjoyed the benefit of early and significant support from colleagues
and management in MCI. Vint Cerf was, as mentioned, one of the early supporters, and so were Teresa Hastings, John Gallant, Bob Spry, and Robert
Oliver who first took the responsibility for developing and deploying SIP in
their respective engineering departments. John Truetken, Lance Lockhart, and
many other engineers in MCI also had critical contributions to the implementation of SIP. Fred Briggs, Patrice Carroll, Barry Zip, and Leo Cyr from MCI
helped with the challenge to develop marketable services based on SIP. We
were fortunate to work jointly in the development and deployment of SIP services with Steve Donovan, Diana Rawlins, Dean Willis, Robert Sparks, Ben
Campbell, Chris Cunningham, Kevin Summers, and many other engineers
from MCI and elsewhere in the industry engaged in the development of SIP in
the Internet Engineering Task Force (IETF).
Most ideas and inspirations driving SIP are due to Prof. Henning
Schulzrinne from Columbia University and to Jonathan Rosenberg from
DynamicSoft and are reflected in this book. Among the many industry contributors, we gratefully acknowledge discussions and guidance from Rohan
Mahy from Cisco Corporation, Gonzalo Camarillo and Adam Roach from
L.M. Ericsson. Jiri Kuthan from GMD Focus, Berlin, was helpful with SIP tutorial charts and with discussions in transatlantic calls using SIP phones—again,
calls of crystal clear clarity to our surprise. The authors are grateful to Richard
Shockey from NeuStar, Inc. and Douglas Ranalli from NetNumber, Inc. for
numerous discussions regarding ENUM. Theodore Havinis has contributed to
the SIP-QoS-AAA aspect for mobile users.
xxiii
xxiv Acknowledgments
We acknowledge countless helpful discussions and insight from many participants in the IETF and especially to Scott Bradner for holding the authors
and others in the IETF SIP community in line to the true conceptual, technical,
and procedural spirit of the Internet.
Jeff Pulver has played a special role in providing a platform and leading
exhibition of products for what was initially an obscure and unknown protocol in the Voice ON the Net (VON) and other conferences held in America,
Europe, and Asia.
Carrol Long, Kevin Shafer, and Adoabi Obi Tulton from John Wiley & Sons
have been instrumental in editing this book.
Introduction
The second edition of Internet Communications Using SIP had to be rewritten
almost from the ground up, because of the dramatic changes in the industry in
the five years that have passed since the first edition. Some of the developments
had been envisaged in the first edition, but naturally, some have not.
The Internet Has Replaced the Telephone System and
the Telecommunication Networks
Since the publication in 2001 of the first edition of this book, Internet Communications Using SIP, Voice over IP (VoIP) has developed from an emerging technology to the recognized replacement of existing global telephone systems
based on Time Division Multiplex (TDM) circuit switching. The Internet has
also replaced the proposed connection-oriented offsprings of TDM, such as the
Integrated Services Digital Network (ISDN) and the Asynchronous Transfer
Multiplex (ATM) based broadband version BISDN, envisaged for the telecommunications industry by the International Telecommunications Union ITU-T
standards body. TDM, ATM, ISDN, and BISDN are now history.
All wired and wireless communications are instead migrating to the Internet
standards developed by the Internet Engineering Task Force (IETF). The legacy
telecommunication networks, while still dominant, are recognized as a presentday cash cow only and are scheduled for replacement by IP networks.
The end-to-end nature of the Internet that places intelligence in the applications running in the endpoints and gives control to the user at the endpoints
has indeed replaced TDM-based telephony with central control. The Internet
xxv
xxvi Introduction
has also proven to be the home network for other types of communications,
information, entertainment, and data applications. To quote Jon Peterson, area
director of the IETF:
“The Internet is the service.”
The Session Initiation Protocol Is the Standard
for VoIP and Multimedia Communications
Another change from the first edition of this book is the Session Initiation Protocol (SIP), which has been adopted by practically all public VoIP service
providers for wired and wireless communications. The discussions about SIP
versus H.323 standardized by the ITU-T are over as well. The installed base of
H.323 is considered a liability and planned for replacement by SIP sooner or
later.
A global industry has emerged to take advantage of SIP and its associated
IETF standards for real-time communications. More than 560 VoIP service
providers have been reported [1] in early 2006, most of them using SIP-based
networks. The list of SIP-based equipment (such as SIP phones, software for
PCs, and mobile devices, servers, gateways, and so on) is now large and still
growing. Actually, all equipment and system vendors are now supporting SIP.
Presence and Instant Messaging Are
Mainstream Communications
Presence and instant messaging (IM) are now mainstream with consumers
and, in the enterprise, complementing or sometimes replacing voice communications in specific situations (such as in circumstances where silence is
required). Even for VoIP, presence has emerged not only as a valuable
enhancement, but presence may be the dial tone of the twenty-first century.
Presence and event-based communications have enabled the integration of
communications with applications. Presence and IM are discussed in Chapter
13, “Presence and Instant Messaging.”
The so-called IM services provided by large Internet companies, such as
AOL, Apple, Google, IBM, Microsoft, Skype (not SIP-based), and Yahoo!, actually carry at present most of the public VoIP traffic between end users around
the globe.
It is not far-fetched to see the IM Internet companies replacing the former
telephone companies in the voice communication business. Many legacy
telecommunication companies are also using VoIP to replace the internal TDM
voice networks, but their VoIP services may not survive the advanced technologies deployed by the IM Internet companies and the challenge posed by
peer-to-peer (P2P) communications.
Introduction xxvii
Redefining Communications: Mobility, Emergency and
Equal Access for the Disabled
Internet communications have been known not to be dependent on the location on the Internet. Application-level mobility based on SIP is a key component to seamless mobile communications, as discussed in Chapter 15, “SIP
Application Level Mobility.”
Emergency calling services by users in distress using the Internet (such as
911 in the United States or 112 in Europe) are far more powerful and cost less
than the Public Switched Telephone Network (PSTN) based emergency services. Internet-based emergency calling is indeed in the design stage in a number of countries. Chapter 16, “Emergency and Preemption Communication
Services,” discusses Internet-based emergency services.
The multimedia nature of Internet communications gives hearing- and
speech-impaired people the opportunity to fully participate in rich communications for work and in personal life. Chapter 17, “Accessibility for the Disabled,” discusses access to communications for disabled people.
The Rise of Peer-to-Peer Communications
P2P traffic has risen in the Internet since around 2000 and became the dominant part of Internet traffic by 2004. Since 2004, Skype (which is based on P2P
VoIP, IM, and presence) has also become by far the dominant VoIP provider
worldwide. Since P2P SIP standards work is just emerging as of this writing,
Skype can be considered a prestandard P2P Internet communication service.
The reasons for the emergence of overlay networks and P2P applications
and their nature are discussed in Chapter 20, “Peer-to-Peer SIP,” and also in
Chapter 6, “SIP Overview.” Though the present VoIP industry is built on
client-server (CS) SIP, this may significantly change. To quote David Bryan
from p2p.org:
“P2P SIP may change VoIP to the same extent that VoIP has changed telecommunications.”
VoIP and Multimedia Communications Services Are Still
Fragmented
In spite of all the technological progress, VoIP, IM, presence, and multimedia
services are still a highly fragmented industry:
■■
Telephone services based on VoIP operate as islands and can interconnect (as of this writing) using mostly the legacy Public Switched Telephone Network (PSTN). The service model is giving broadband users
xxviii Introduction
access to the legacy telephone system, actually a voice gateway service
between the Internet and TDM. The business model of most VoIP service providers is just lower cost for legacy-style telephone service, also
called PSTN over IP. The PSTN gateway services are using IP inside
their networks, but users are not exposed to the rich IP services, except
when all parties are on the same network.
■■
The most successful public voice, IM, and presence service is Skype,
which is not standards-based.
■■
Walled gardens: The fragmentation of communications is still actively
pursued by most mobile service providers by deploying systems where
their users can get rich IP multimedia services only on their own networks. The fees to communicate between mobile service providers are a
significant part of the business model, and open connectivity to the
Internet (“Internet neutrality”) is still a hotly debated issue. Internet
neutrality is also still debated by many broadband Internet access
providers (such as DSL and cable companies), although we believe that
enlightened government regulators in the developed countries will
weigh in favor of users and open network access in general.
The proliferation of islands for communications makes them less useful the
more there are, since this proliferation is in denial of Metcalf’s law that the
value of a network increases with the square of the number of points attached
to the network. The Internet with more than 1 billion attached endpoints has
thus the highest value for communications. By contrast, the mobile phone
industry boasts 3 billion users, but in many fragmented networks.
Past Obsessions and Present Dangers: QoS and Security
Network-based quality of service (QoS) for voice and the reliability of the
legacy telephone network have long been used by telephone industry marketers to scare users away from VoIP. In the meantime, all public VoIP services
have proven that Internet best-effort QoS works just fine, as long network congestion is avoided. Internet-based voice can actually be much better than the
3.1 kHz voice over the PSTN. As for reliability, all recent major man-made and
natural disasters have proven the Internet and VoIP to be more resilient than
the existing wireline and wireless telephone networks.
Chapter 18, “Quality of Service for Real-Time Internet Communications,” is
aimed at a balanced approach for QoS, and Chapter 16, “Emergency and Preemption Communication Services,” discusses the Emergency Services based
on SIP.
Introduction xxix
The security threats on the Internet have provided well-justified concerns
about the security of VoIP, and even more, the security of IM. As a result, a new
industry niche, that of VoIP and IM security, has sprung up and, as usual, marketers are first drumming up the vulnerabilities of Internet communications to
prepare the sell for all kinds of security products. Though no significant security breaks have been reported so far for Internet communications, security for
VoIP and IM is still work in progress. Chapter 9, “SIP Security,” deals with SIP
security.
References
[1] A list of VoIP companies is provided at www.myvoipprovider.com.
CHAPTER
1
Introduction
The telecommunications, television, and information technology (IT) network
industries are all transformed by the Internet. The transformation is driven by
the need for growth based on new services, more complete global coverage,
and consolidation. In this chapter, we will explore some of the problems and
solutions for end users and every type of business because of the profound
disruptions caused by the Internet.
Problem: Too Many Public Networks
Before the emergence of the Internet, users and service providers were generally accustomed to thinking in terms of four distinct network types: Networks
for IT (data), networks for voice, mobile networks, and networks for television. Each of these dedicated network types could, in turn, be divided into
many incompatible regional and even country-specific flavors with different
protocol variants.
Thus, we find many types of telephony numbering plans, signaling, and
audio encodings; several TV standards; and various types and flavors of what
the telecom industry calls data networks—all of them incompatible and impossible to integrate into one single global network.
1
2
Chapter 1
The mobile telephone networks have converged on a smaller number
of standards in the second generation (2G) networks and in the emerging third
generation (3G) mobile networks. It may turn out, however, that with the
proliferation of new radio technologies for the so-called 4th generation (4G),
such as Wi-Fi and WiMAX, all modern mobile networks will become just a
wireless access mechanism to the Internet, where all public communications,
entertainment, and applications will reside anyhow.
Data networks that originated in the telecom industry came in many forms,
such as digital private lines, X.25, Integrated Services Digital Network (ISDN),
Switched Multimegabit Data Service (SMDS), Frame Relay, and Asynchronous
Transfer Mode (ATM) networks. These so-called data networks were mostly
inspired by circuit-switched telephony concepts. Their names are meant to
suggest that they were not designed primarily to carry voice.
Voice networks are still used for data and fax because of their general availability, though less and less so. However, these networks have come to the end
of their evolution, since they are fundamentally optimized for voice only. TV
networks were designed and optimized for the distribution of entertainment
video streams.
Needless to say, all network types (data, voice, TV, and mobile) have specific
end-user devices that cannot be ported to other service providers or network
types, and most often cannot be globally deployed.
The impact of the Internet has made the wired and wireless phone companies and the TV cable companies look for new business models that can take
advantage of Internet technologies and protocols, among them the Session Initiation Protocol (SIP) for real-time communications, such as Voice over IP
(VoIP), instant messaging (IM), video, conferencing/collaboration, and others.
Examples of the various categories and their business models are illustrated in
Table 1.1. We assume that most readers are familiar with the acronyms used in
the table, and we also explain these acronyms and terms in the book. They can
also be found in the index.
Table 1.1
Internet Communications in 2005 with Examples from North America
CATEGORY
WHO
PROTOCOLS
Open IM services Pulver FWD, Standard SIP
with VoIP voice
Gizmo/
(competing islands) SIPphone,
Damaka,
Ineen
STRENGTHS
WEAKNESSES
Internet
Presence
Video
User gets SIP
URI
On Net is free
Limited
financing
Introduction
CATEGORY
WHO
PROTOCOLS
STRENGTHS
WEAKNESSES
Closed IM islands
with VoIP
Yahoo, MSN, SIP or other
Google, AOL,
Skype
(the most
innovative)
Internet
Nonstandard
Presence
Walled gardens
Video
On Net is free
PSTN gateways
PSTN over IP
Most “VoIP” SIP
companies
Internet
Anywhere
Video (Packet8)
On net is free
Low-cost PSTN
No new services
Compete on
price
Costly
infrastructure
Telephony
over cable
TV cable
companies
Everything from
PSTN to MGCP
to SIP with “P-”
extensions
Broadband
Internet
Access to
80%+
households
Large
investments in
PSTN and older
VoIP flavors
Wireless walled
gardens
3G mobile
operators
SIP for IMS
with “P-”
extensions
Strong
financing
Wireline emulation Wireline
of IMS: TISPAN
phone
companies
“NGN”
SIP with
“P-”
extensions
Central control
inhibits
innovation
IP network cost
Duplicate IMS &
NGN services
The proliferation of isolated communication islands as shown in Table 1.1
makes them less useful as their number keeps increasing (think of many more
communication islands all over the world). Building communication islands
(also called “walled gardens”) is in conflict with Metcalfe’s law that the value
of the network increases by the square of the number of connected endpoints.
Last, but not least, in case of an emergency, having many networks that cannot
communicate directly is not very helpful.
Closed networks are an impediment for innovation, since innovators must
work (technology and legal agreements) with every closed network separately
to bring a new service or product to market. By contrast, the Internet extends
the reach for new applications and services instantly to the whole world.
3
4
Chapter 1
Another observation from Table 1.1 is that the strongest financing available
is at present for closed networks (walled gardens), the ones that are most limited in reach and usefulness. This raises business issues and regulatory questions (what are the public interest obligations, if any?) that are beyond the
scope of this book.
Incompatible Enterprise Communications
Enterprise communication systems are often an even greater mix of incompatible and disjoint systems and devices:
■■
Proprietary PBX and their phones. Phones from one PBX cannot be
used by another.
■■
Instant messaging is a separate system from the PBX.
■■
Various IM systems don’t talk to each other.
■■
Voice conferencing and web-based collaboration use yet other systems.
Maintaining various incompatible and nonintegrated proprietary enterprise
systems is quite costly and reduces the overall productivity of the workforce.
Network Consolidation: The Internet
The Internet has benefited from a number of different fundamentals compared
to legacy networks, such as the tremendous progress of computing technology
and the open standard Internet protocols that define it. This progress can be
attributed to the expertise of the research, academic, and engineering communities whose dedication to excellence and open collaboration on a global basis
have surpassed the usual commercial pressure for time-to-market and competitive secrecy.
The result is an Internet that uses consistent protocols on a global basis, and is
equally well suited to carry data, transactions, and real-time communications,
such as instant messaging (IM), voice, video, and conferencing/collaboration.
Actually, the Internet is the “dumb network,” designed for any application,
even those not yet invented. This is in stark contrast to the isolated “walled
gardens” with central control of all services illustrated in Table 1.1.
Introduction
Voice over IP
Although the Internet has quickly established itself as the preeminent network
for data, commercial transactions, and audio-video distribution, the use of
voice over the Internet has been slower to develop. This has less to do with the
capability of the Internet to carry voice with equal or higher quality than the
telephone network but rather with the complex nature of signaling in voice
services, as you will see in Chapter 6, “SIP Overview.”
There are various approaches for voice services over the Internet, based on
different signaling and control design. Some examples include the following:
■■
Use signaling concepts from the telephone industry—H.323, MGCP,
MEGACO/H.248.
■■
Use control concepts from the telephone industry—central control and
softswitches.
■■
Use the Internet-centric protocol—Session Initiation Protocol (SIP), the
topic of this book.
The movement from such concepts as telephony call models to discovery/rendezvous and session setup between any processes on any platform
anywhere on the Internet is opening up completely new types of communication services.
The use of SIP for establishing voice, video, and data sessions places telephony as just another application on the Internet, using similar addressing,
data types, software, protocols, and security as found, for example, on the
World Wide Web or e-mail.
Separate networks for voice are no longer necessary, and this is of great consequence for all wired and wireless telephone companies.
Complete integration of voice with all other Internet services and applications probably provides the greatest opportunity for innovation. The open and
distributed nature of this service and the “dumb” network model will
empower many innovators, similar to what has happened with other industries on the Internet and the resulting online economy.
Most IM systems on the Internet already have voice and telephony capability as well, though if it is proprietary, they cannot intercommunicate without
IM gateways, although IM gateways inevitably cannot translate all the
features from one system to another. IM gateways are also transitory in nature,
5
6
Chapter 1
since any changes to a proprietary IM protocol may render the gateway close
to useless. By contrast, SIP-based communications offer a global standardsbased approach for interoperability for presence, IM, voice, and video, as we
will show in the following chapters.
Presence—The Dial Tone for the Twenty-First
Century?
Unsuccessful telephone calls are a serious drag on productivity and a source of
frustration, since both parties waste time and talk to voicemail instead to each
other. Also, the timing of the phone call may not be appropriate or not reach
the called party in a suitable location. The advent of presence, so well-known
from IM systems, can provide much more rich information before trying to
make a call in the first place, compared to just hearing the dial tone. Another
convenience of SIP and presence is that many contact addresses may reside
beneath a buddy icon, so the caller need not to know or worry about picking
the right phone number or URI. Presence may, therefore, replace the dial tone
used in telephony for well over 100 years.
The Value Proposition of SIP
SIP is not just another protocol. SIP redefines communications and is impacting
the telecom industry to a similar or greater degree than other industries. This has
been recognized by all telecom service providers and their vendors for wired
and wireless services, as well as by all IT vendors. Chapter 2 will provide an
overview of how the Internet and SIP are redefining communications.
SIP Is Not a Miracle Protocol
As discussed in Chapter 2, “Internet Communications Enabled by SIP,” SIP is
not a miracle protocol and is not designed to do more than discover remote users
and establish interactive communication sessions. SIP is not meant to ensure
quality of service (QoS) all by itself or to transfer large amounts of data. It is not
applicable for conference floor control. Neither is it meant to replace all known
telephony features, many of which are caused by the limitations of circuitswitched voice or to the regulation of voice services. And such a list can go on.
Various other Internet protocols are better suited for other functions. As for
legacy telephony, not all telephone network features lend themselves to replication on the Internet.
Introduction
The Short History of SIP [1]
By 1996, the Internet Engineering Task Force (IETF) had already developed the
basics for multimedia on the Internet (see Chapter 14, “SIP Conferencing”) in
the Multi-Party, Multimedia Working Group. Two proposals, the Simple Conference Invitation Protocol (SCIP) by Henning Schulzrinne and the Session Initiation Protocol (SIP) by Mark Handley, were announced and later merged to
form Session Initiation Protocol. The new protocol also preserved the HTTP
orientation from the initial SCIP proposal that later proved to be crucial to the
merging of IP communications on the Internet.
Schulzrinne focused on the continuing development of SIP with the objective of “re-engineering the telephone system from ground up,” an “opportunity that appears only once in 100 years,” as we heard him argue at a time
when few believed this was practical.
SIP was initially approved as RFC [2] number 2543 in the IETF in March
1999. Because of the tremendous interest and the increasing number of contributions to SIP, a separate SIP Working Group (WG) was formed in September
1999. The SIP for Instant Messaging and Presence Leveraging (SIMPLE) was
formed in March 2001, followed by SIPPING for applications and their extensions in 2002. The specific needs of SIP developers and service providers have
led to an increasing number of new working groups. This very large body of
work attests both to the creativity of the Internet communications engineering
community, and also to the vigor of the newly created industry.
We will shorten the narrative on the history of SIP by listing the related
working groups (WG) in chronological order in Table 1.2. We have listed for
simplicity the year of the first RFC published by the WG, though the WG was
sometimes formed one to two years earlier. Years denote a new WG that has
not yet produced any RFC.
Table 1.2
History of SIP-Related Working Groups
NAME
FIRST RFC
CHARTER
avt
1996
Real-time transmission of audio and video over
UDP/IP: RTP
mmusic
1998
Internet conferencing and multimedia
communications: SIP, SDP, RTSP
iptel
2000
Routing and call processing for IP telephony: TRIP,
CPL, tel URI
sip
2000
Development of the SIP protocol: SIP methods,
messages, events, URI
(continued)
7
8
Chapter 1
Table 1-2
(continued)
NAME
FIRST RFC
CHARTER
enum
2000
DNS-based use of ITU-T E.164 telephone numbers
sipping
2002
Applications and extensions to SIP
simple
2004
Use of SIP for Instant Messaging (IM) and
Presence
xcon
2005
Centralized conferences
behave
(2005)
Behavior for Network Address Translation (NAT) for
use with SIP, RTP
ecrit
(2005)
Emergency communications (such as 911, 112)
p2psip
(2005)
Peer-to-peer SIP (not yet a formal WG)
The growth of SIP-related standards in the IETF is illustrated and discussed
in Chapter 21, “Conclusions and Future Directions.”
References in This Book
Because of the multiple developments on the Internet, SIP is being used in
ever-more services, user software, and various user devices (such as in SIP
phones, PCs, laptops, PDAs, and mobile phones). This is, in effect, a new
industry and its participants keep making new contributions to the core SIP
standards, mainly in the area of new services and new applications. This book
reflects SIP developments up to and including the 64th IETF in November
2005.
We have included, by necessity, many Internet drafts that are designated
work in progress, since they are the only reference source for this particular
information. Some of these drafts may become standards by the time you are
ready to use them; some may be a work in progress and have a higher version
number than quoted as of this writing; and still others may be found only in an
archive for expired drafts.
The SIP WG drafts that are work in progress can be found online at the IETF
web site:
http://ietf.org/html.charters/sip-charter.html
Additional individual submissions and Internet drafts from other working
groups can be found at the following site:
http://ietf.org/ID.html
Introduction
SIP-related drafts that have expired (older than six months) can be found on
several archives. As of this writing, following are some of the sites:
www.cs.columbia.edu/sip/drafts
www.softarmor.com/sipwg
Readers may also perform a web search, such as Google, for any IETF SIPrelated topic or for any Internet draft or RFC.
Several books have been published on Internet multimedia, Voice over IP,
and SIP, some of which are listed here. [3], [4], [5] They focus mainly on how
SIP works. This book is less about explaining how SIP works, but rather what
it does and the new communications and services it enables.
We have reproduced some of the exciting services and features discussed in
the IETF SIP WG and its main offsprings, the SIPPING and SIMPLE Working
Groups. Also in